Has anyone experience of using Internet phone (Skype, Messenger, BT's VOIP)
over a very long line?
My line is claimed to be 8.6Km and I'm getting 480Kbit/s download
synchronisation (it's gone poor recently) with my BT Home Hub which will
translate to 256Kbit/sec IP.
I'm considering DECT phones round the house and am wondering if the extra
VOIP functionality is worth it.

Phil B

On 2007-03-20, Phil B <phil.remove.brady@hotmail.co.uk> wrote:
The speed you are getting upstream will matter too. I've used Skype over a
cable modem service running at 250 kbps downstream and 54 kbps upstream
and it works fine for phone calls if there isn't too much competition for
upstream bandwidth (though Skype to another PC won't). I suspect the
SIP services will do as well, the bandwidth requirements are similar
since they use the same codec as Skype.

Dennis Ferguson

Phil B wrote:

Before you invest in any hardware, invest a little time in setting up a
softphone [X-Lite for example] with a free SIP account from the likes of
Sipgate and make some test calls. That'll give you a better idea of call
quality than anyone in this newsgroup will be able to. Note that if you
intend to use analogue DECT handsets with Skype or Messenger, your PC will
need to be on to make/receive calls.

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"Dennis Ferguson" <dcferguson@pacbell.net> wrote in message
news:slrnf00hqo.80.dcferguson@akit-ferguson.com...

Skype doesn't have a very low bitrate codec does it?
I have used an experimental codec that actually worked over a dial up
connection at about 2kb/s.
You could tell what was being said but it didn't sound human.
That was when we were discussing VoIP and broadband ADSL and SIP didn't
exist.

On 2007-03-21, dennis@home <dennis@killspam.kicks-ass.net> wrote:
Skype uses a G.729 codec for calls to phones. This is the most advanced
(i.e. lowest bit rate) codec that standard SIP services use, at 8 kbps.

That sounds like a small number, and it would be if all you had to send
was the codec data. The thing is, however, that G.729 generates a
compressed sample every 10 milliseconds (8 bytes of data), and Skype
sends it that way, generating 100 packets per second. While each
packet only carries 8 bytes of codec data, it also has to have a 20
byte IP header and maybe 8 or 12 bytes of transport header, plus a few
bytes for a link layer header and CRC, so 8 kbps of codec data becomes
closer to 40 kbps by the time you put it on the wire. The SIP services
aren't that much different. I think Skype might be able to drop back
to 50 packets per second (16 bytes of codec data per packet) if it
detects that it is running over a slow circuit, but that still doesn't
get it much under 30 kbps. I've heard people say Skype will run over
a dialup modem, but if it does it is using all of the upstream bandwidth.

The codec matters less than one might think for VoIP since most of what
the IP packets carry isn't codec data. It is IP packet overhead. Since you
use up too much of your delay budget in packetization if you transmit
less than about 50 packets per second, even a 2 kbps codec would still
add up to a pretty high data rate by the time it got to the wire.

Skype's PC-to-PC codec, as I understand it, is a non-standard 16 ksps,
16 bit sample codec. It seems to generate 50 packets per second. While
it is a variable rate codec, it peaks at over 80 kbps on the wire, which is
why it wouldn't work over my cable modem service.

Dennis Ferguson

Dennis Ferguson wrote:

Looking at those numbers RTP could make all the difference for DSL
(Connecting pppoa/vc mux) without it 42400 bit/s with 84800. I never use
skype or voip, so don't know whether you can ask it not to use rtp.

It doubles because you can get 10 bytes in a udp packet and it will fit
in one atm cell, one byte extra needs 2 cells and rtp adds 12.

Andy.